Techniques for amplification of an analog signal, sampling and converting the signal to digital and processing that signal using digital techniques are known in the art.
Instrumentation amplifiers are commonly used to amplify values of an analog signal. Noise, distortion and offset are critical performance parameters.
Following an instrumentation amplifier in a signal processing chain is typically an analog to digital converter. At its input, the signal is sampled onto a capacitor. To reduce loading effects of the sample process used to sample an analog signal, a rough buffer may be used to precharge the sampling capacitor followed by a period of fine adjustment. The sampled analog signal is converted to digital, such as a one bit digital stream and filtered to produce a multibit digital signal.
Filters for doing such processing, such as FIR filters and FIR sinc filters are known. Some such filters may use coefficients for multiplying digital values. Others, such as Hogenauer filters, described in an article by Eugene B. Hogenauer, entitled “AN ECONOMICAL CLASS OF DIGITAL FILTERS FOR DECIMATION AND INTERPOLATION,” published in IEEE Transactions on Acoustics, Speech and Signal Processing, Volume ASSP-29, No. 2, April 1981, perform the filtering without coefficients.
A typical data acquisition system may consist of an ADC preceded by signal conditioning circuitry and followed by digital signal processing and communication circuits. Often the digital signal processing circuitry includes an FIR filter, typically performing the function of decimation and low pass filtering of the signal.
Any FIR filter will have a settling time. The function of the FIR can be described asY(z)=Sum(Ai*X*z−i)
where Ai are the coefficients of the filter, X is the input, Y the output and i the index of the taps running from 0 to n. X*z−i then represents the data delayed by i clock cycles.